| /external/webrtc/webrtc/common_audio/signal_processing/ |
| get_scaling_square.c | 37 t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
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| auto_correlation.c | 38 int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
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| signal_processing_unittest.cc | 40 EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
41 EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| get_lsp_poly.c | 56 (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
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| frame_classify.c | 79 (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
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| smooth.c | 103 WEBRTC_SPL_MUL(ENH_A0, w00prim >> 14),
184 WEBRTC_SPL_MUL(A, w11_div_w00);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| bandwidth_estimator.c | 301 bweStr->recBwInv = WEBRTC_SPL_MUL((int32_t)bweStr->recBwInv, (int32_t)reductionFactor);
394 tempUpper = WEBRTC_SPL_MUL(tempUpper, numBytesInv);
395 tempLower = WEBRTC_SPL_MUL(tempLower, numBytesInv);
431 arrTimeProj = WEBRTC_SPL_MUL((int32_t)8000, recBwAvgInv);
435 arrTimeProj = WEBRTC_SPL_MUL(((int32_t)pksize + HEADER_SIZE), arrTimeProj);
468 WEBRTC_SPL_MUL(973, bweStr->recJitterShortTermAbs);
474 WEBRTC_SPL_MUL(3891, bweStr->recJitterShortTerm);
512 bweStr->recMaxDelay = WEBRTC_SPL_MUL(3, bweStr->recJitter);
566 bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
574 bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg)
[all...] |
| entropy_coding.c | 206 sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
224 sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
280 sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
298 sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
426 dither_gain_Q14 = (int16_t)(22528 - WEBRTC_SPL_MUL(10, AvgPitchGain_Q12));
[all...] |
| arith_routines_logist.c | 73 ind = WEBRTC_SPL_MUL(5, qtmp1 - kHistEdges[0]);
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| lattice.c | 22 ((int32_t)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
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| lattice_neon.c | 187 *ptr2 = (int32_t)(WEBRTC_SPL_MUL(t16a, tmp32b) +
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| isacfix.c | [all...] |
| /external/webrtc/webrtc/common_audio/signal_processing/include/ |
| signal_processing_library.h | 39 #define WEBRTC_SPL_MUL(a, b) \
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| entropy_coding.c | 107 sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
127 sum += WEBRTC_SPL_MUL(ARCoefQ12[n - k], ARCoefQ12[n]); /* Q24 */
[all...] |
